Hey ffmpeg Team,
I am trying to convert a wav file to an amr wideband format. I tried
using different encoders:
ffmpeg -i test/calibre.wav -f awb -c:a libvo_amrwbenc calibre.awb
or:
ffmpeg -i test/calibre.wav -f awb -c:a libfdk_aac calibre.awb
I also tried without the "-f awb" option.
When typing:
ffmpeg -i test/calibre.wav -c:a libvo_amrwbenc calibre.awb
I get the error:
ffmpeg version N-114151-g4700925d22 Copyright (c) 2000-2024 the FFmpeg
developers
built with gcc 11 (Ubuntu 11.4.0-1ubuntu1~22.04)
configuration: --enable-libvo-amrwbenc --enable-libfdk-aac
--enable-libgsm --enable-libmp3lame --enable-libopencore-amrwb
--enable-libopencore-amrnb --enable-version3
libavutil 59. 1.100 / 59. 1.100
libavcodec 61. 1.101 / 61. 1.101
libavformat 61. 0.100 / 61. 0.100
libavdevice 61. 0.100 / 61. 0.100
libavfilter 10. 0.100 / 10. 0.100
libswscale 8. 0.100 / 8. 0.100
libswresample 5. 0.100 / 5. 0.100
[aist#0:0/pcm_s16le @ 0x5647e88f1a40] Guessed Channel Layout: stereo
Input #0, wav, from 'test/calibre.wav':
Metadata:
title : Calibre - Feeling Normal - 03 Time To Breathe
(with Cimone)
NITR : NTKB?y
Duration: 00:05:50.50, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz,
stereo, s16, 1411 kb/s
[AVFormatContext @ 0x5647e8933d00] Unable to choose an output format
for 'calibre.awb'; use a standard extension for the filename or
specify the format manually.
[out#0 @ 0x5647e8933bc0] Error initializing the muxer for calibre.awb:
Invalid argument
Error opening output file calibre.awb.
Error opening output files: Invalid argument
When I look for the "ffmpeg -encoders" available it says, that they
should be availabe:
A....D libopencore_amrnb OpenCORE AMR-NB (Adaptive Multi-Rate
Narrow-Band) (codec amr_nb)
A....D libvo_amrwbenc Android VisualOn AMR-WB (Adaptive
Multi-Rate Wide-Band) (codec amr_wb)
I hope you can help me with that.
Best regards
Ecki
You need to set the bitrate with -b and if your input has a samplerate
that is not 16000 Hz and is not mono, you need to set -strict to
"unofficial"
Your input is stereo and probably has a higher samplerate, so
unsupported by the codec unless unofficial is set. You also dit not set
a supported bitrate.
It is all in the manual here:
https://ffmpeg.org/ffmpeg-codecs.html#toc-libvo_002damrwbenc
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